Hey everyone --
Sorry about the multiple postings, but I figured what the heck...
I've just put on-line a whole lot of work I've done; papers, pieces,
software, etc. Here's the link for the 2-3 people (beyond my immediate
family) who might be interested:
There's a fair amount of unix/linux work scattered throughout, including
the big "My Music Book" thing I did a few years ago.
Hope you enjoy this!
> I'm hoping that you're thinking of a realtime display, in which the
> peaks roll off to create a true waterfall effect.
Baudline (http://www.baudline.com) is a fantastic viewer that does fft
cascade. I've used it for a couple of years, and it is great for figuring out
how different sounds "work", and it has an oscilloscope-type display as well.
I finally started making my pet music project and realized I need a
drum synth to make some cool sounds. psindustrializer is good but also
need some tr-909-style sounds. I remeber from my old windoze days I
used a nice piece of software called Stomper. Does anybody know any
software for linux with comparable capabilities? Or we need to write
Stomper does not work under wine :(
I had a couple of articles on drum synths. Check
I built the circuit in a00*.jpg at the time when this article
was fresh. The article b00*jpg mentions an earlier article.
I will check that out at library.
Hmm.. I coded a drum synth for Commodore VIC-20 at the time.
VIC provided an audio chip with three oscillators, noise,
and a common volume if I remember correctly. What I did was to
modulate osc pitch and volume parameters with a fast and accurate
(compared to Basic) assembly code. The drum sounds were assigned to
the keys. This was about 1984, inspired by Yamaha's digital RX drum
synths, not by analog drums.
for developers of open source graphics software
Ladies and Gents,
I can hack specimen while working full time. And, I can hack specimen
while studying full time. But, and this is empirically verifiable, I
can't hack specimen while both working and studying full time. And my
situation is not likely to change for another year or so.
What this means is that I'm just not cut out to run a project right now,
unless I want to put it into maintenance mode. Really, that's where
specimen has been for the past six months anyway, and I've been doing a
rather piss poor job at that modest role! My efforts are better applied
to tasks where smaller chunks of time can go a greater distance.
I don't have any regrets --- this was my first real programming project,
and I learned a lot. But the truth is that LMMS is more specimen than
specimen right now, and it has an active development and user community
surrounding it. Plus, I've always been a musician and an artist first,
and a programmer second.
All told, it's time to throw in the towel on this one. In a way, this
is a bummer --- I've put a lot of sweat and tears (literally) into this
project over the past couple of years, and it has come an incredible
distance. But I'd be a fool to think that I'm better off keeping it
afloat than making music and contributing to other projects.
And truthfully, it's a huge relief to get this announcement out. A
sense of obligation is what kept me from making it sooner, but in
retrospect, that's pretty ridiculous. Considering that I'm an "open
source, just for fun" guy, and not a "free software, as in freedom"
type, it doesn't make a lot of sense to keep going when pain exceeds
This isn't the end of my open source music development, however. I hope
to help take LMMS to the next level, and contribute to other projects
that will help make Linux a competitor in the music industry. Things
like ardour2, lash, jack-midi, vst, dssi, ladspa et al are the keys to
our future in this regard. And I look forward to getting back to
hardcore hacking in a few years, when I've got my life settled down and
I'm not putting in 80 hours of work and school a week.
Take care everybody, and may the funk be with you.
"So this baby seal walks into a club."
I'm trying to find a library or code-snippet in order to do audio
resampling from 8khz to 44,1khz and from 44,1khz to 8khz. I need to
resample the data in realtime - resampling a buffer of data, not a
soundfile. The quality doesn't need to be good so I guess the best
solution might be linear audio resampling. The device to do the
resampling on is an ARM CM-X255 running at 400MHz.
I tried out libsamplerate so far but when I tested it with the
soundfile conversion test program it needed 3,5 secs to sample from
8kHz to 44,1 khz for a 1,7 secs audiofile - which is too slow for me.
Is there something faster that can do the job?
Any suggestions are highly welcome
First, thank you to everyone who responded to my original query. I now
have a much better idea how I want to do this project.
I definitely want to involve members of this group. Some of you have
offered to take on particular chapters, and that seems to me the best
approach. I believe we can complete the work in record time by spreading
the work around. I'll speak with my publisher about this plan. If he
approves this approach we'll start work immediately.
Some respondents wrote to me off-list and have offered to assist. I'll
be in touch with you all after talking with Bill Pollock. If Bill gives
us a green light I'll start organizing the distribution of work
immediately. I'll communicate directly with you regarding style
preferences, format requirements, deadlines, etc. I must emphasize that
this project is commercial work, i.e., we'll have editors at the
publishing house and there will be demands that certain dates be met.
Please, only take on the work if you're sure you can work under those
If the community approach is accepted I'll post a list of topics and
areas that need attention. I have a large amount of unedited/incomplete
material that I'm willing to offer as starting points.
I've been thinking about the royalties issue. I'm far from a final plan,
but I think it might be possible to route some (all?) of the royalties
to the Linux Audio Consortium. The community can then decide how to use
it. I can't leave this issue vague, my publisher will need to know who
is in charge of accounts receivable, and I can't simply accept
contributions from the community and not reimburse those contributors.
How that reimbursement takes place can be left to the individual
contributors, or we may decide to simply throw everything in one
direction (e.g. linuxaudio.org).
Btw, I especially appreciate the community's response re: the 2.4 kernel
series. It is laid to rest, there will be coverage only from kernel 2.6
upwards. Thank goodness.
No Starch has already accepted my outline for the project. At some point
I'll post that outline on the Web for the community to consider.
However, please bear in mind that there will be limitations of space and
scope. I'm unlikely to be able to cover games, consumer devices, or
telephony. Alas, material for a programmer's guide will probably not be
accepted. IMO that really requires a separate book now. A developer's
guide to ALSA and JACK is certainly timely and needed, but I'm not the
one to write that book.
Again, if you're interested in writing for this project please contact
me off-list. Specify your area of interest and we'll go from there.
Hi to everybody, I am new to this list. Many thanks to all the
developers, It's already more than two years that I'm a happy user of
free software for music on my ibook.
But only recently I've decided that it's worth to learn woh to compile
and install by myself the softwares that I use more often.
I'm using a lot the LADSPA plugins packaged for os x, but I'm trying
to compile the ladspa sdk. I've modified the makefile following the
patch provided with fink, and the compilation seems to go well.
However, at the end, I get the following error:
time ../bin/applyplugin -s 1 \
../snd/noise.wav /tmp/test.wav \
../plugins/filter.so lpf 500 \
../plugins/filter.so lpf 500 \
../plugins/sine.so sine_fcaa 6000 \
../plugins/delay.so delay_5s 1 0.1 \
../plugins/amp.so amp_mono 4 \
Unable to find label "lpf" in plugin library file "../plugins/filter.so".
0.01 real 0.00 user 0.00 sys
make: *** [/tmp/test.wav] Error 1
Can someone help me to solve this problem? I'm using mac os 10.4.4 on
a g4 ibook.
My publisher, Bill Pollock, has been gently pressuring me to commit to
completing the 2nd edition of The Book Of Linux Music & Sound.
Unfortunately I'm in a precarious position to commit myself to the work.
The first book nearly wiped me out, I'm not sure I can sustain the
effort to bring the next edition to light. Nevertheless, I'm still
interested in seeing this book through to completion. So I have some
questions for the community :
1. Is there a real need for another book such as the The Book Of Linux
Music & Sound ?
2. If so, would I be wise to ignore the 2.4 kernel series ? (It would
make it easier to ignore material re: OSS/Free)
3. Would anyone be interested in co-authoring the book ? I've
considered offering some chapters to certain people on these lists, but
the issue of reimbursement gets sticky WRT royalties and other
compensation. I made very little money from the first book, but money
wasn't the true reward anyway, so perhaps there's a way to turn it into
a community-based work.
4. Is anyone else already working on such a project ? I don't want to
Btw, this is the last hurrah for this project. If I don't take it now
I won't be taking it on at all. I have a life, it's pretty full, and
committing to this edition would be a major disruption. I can guarantee
that it would be the last book I'll ever write.
I look forward to your comments and advice.
> Hello List,
> I'm trying to find a library or code-snippet in order to do audio
> resampling from 8khz to 44,1khz and from 44,1khz to 8khz. I need to
> resample the data in realtime - resampling a buffer of data, not a
> soundfile. The quality doesn't need to be good so I guess the best
> solution might be linear audio resampling. The device to do the
> resampling on is an ARM CM-X255 running at 400MHz.
> I tried out libsamplerate so far but when I tested it with the
> soundfile conversion test program it needed 3,5 secs to sample from
> 8kHz to 44,1 khz for a 1,7 secs audiofile - which is too slow for me.
> Is there something faster that can do the job?
Was this with the linear resampler lib libsamplerate? In case, you should
know that its terrible slow. Last time I tried, the fastest sinc resampler
in CLM was almost as fast as the linear resampler in libsamplerate.
(Erik, you should do something about that...)
I don't know of any other libraries that does linear resampling, but its
not that difficult to make one manually. I'm sure there are example codes
floating around as well.